[asterisk-users] NAME/USERNAME conflict

[kazabe]

Hi.

we are experimenting a strange issue in our PBX.

By example: if we dial to the 100, the call is answered in 199.  We dont
have any redirection for that, but the cli show the same issue when request
show peers.  Aditionally, the user 100 use the ip address 192.168.11.100,
and the cli show connected the user from 192.168.11.160 (that ip is
assigned to the user 199)

PBX*CLI> sip show peers
Name/username             Host                                    Dyn
Forcerport Comedia    ACL Port     Status      Description
100/199                   192.168.11.160                           D  Yes
     Yes         A  5060     OK (30 ms)


I check the sip 100 and (aparently) show all normal


PBX*CLI> sip show peer 100


  * Name       : 100
  Description  :
  Secret       : <Set>
  MD5Secret    : <Not set>
  Remote Secret: <Not set>
  Context      : MAIN
  Record On feature : automon
  Record Off feature : automon
  Subscr.Cont. : <Not set>
  Language     :
  Tonezone     : <Not set>
  AMA flags    : Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup    : 1
  Pickupgroup  : 1
  Named Callgr :
  Nam. Pickupgr:
  MOH Suggest  :
  Mailbox      : 100 at device
  VM Extension : *97
  LastMsgsSent : 0/0
  Call limit   : 2147483647
  Max forwards : 0
  Dynamic      : Yes
  Callerid     : "JDOE" <100>
  MaxCallBR    : 384 kbps
  Expire       : 2680
  Insecure     : no
  Force rport  : Yes
  Symmetric RTP: Yes
  ACL          : Yes
  DirectMedACL : No
  T.38 support : No
  T.38 EC mode : Unknown
  T.38 MaxDtgrm: 4294967295
  DirectMedia  : No
  PromiscRedir : No
  User=Phone   : No
  Video Support: No
  Text Support : No
  Ign SDP ver  : No
  Trust RPID   : Yes
  Send RPID    : No
  TrustIDOutbnd: Legacy
  Subscriptions: Yes
  Overlap dial : Yes
  DTMFmode     : rfc2833
  Timer T1     : 500
  Timer B      : 32000
  ToHost       :
  Addr->IP     : 192.168.11.160:5060
  Defaddr->IP  : (null)
  Prim.Transp. : UDP
  Allowed.Trsp : UDP
  Def. Username: 199
  SIP Options  : path replaces replace timer
  Codecs       : (ulaw)
  Codec Order  : (ulaw:20)
  Auto-Framing : No
  Status       : OK (28 ms)
  Useragent    : Grandstream GXP2000 1.2.5.3
  Reg. Contact : sip:101 at 192.168.11.160:5060;transport=udp
  Qualify Freq : 60000 ms
  Keepalive    : 0 ms
  Sess-Timers  : Accept
  Sess-Refresh : uas
  Sess-Expires : 1800 secs
  Min-Sess     : 90 secs
  RTP Engine   : asterisk
  Parkinglot   :
  Use Reason   : No
  Encryption   : No

What can cause that?  i delete both extensions and create again and the
problem continue.  Adn others extensions are showing the same issue (call
to another extension and answer at 199).

thanks in advance
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